In recent years, for speech signal and audio signal coding, scalable coding techniques have been developed whereby speech and audio signals can be decoded from a portion of encoded information to reduce sound quality deterioration even under conditions in which packet loss occurs (for example, see Patent Document 1). With these scalable coding techniques, it is possible to decode speech and audio signals from a portion of encoded information to reduce sound quality deterioration even under conditions in which packet loss occurs. To be more specific, one representative example is a method of repeating: encoding an input signal and generating encoded information of the first layer; generating in the (i−1)-th layer representing the higher layer (i is an integral number equal to or greater than 2), a residual signal showing the difference between the input signal and a decoded signal acquired according to encoded information of the (i−1)-th layer; and performing coding according to a residual signal in the i-th layer representing the much higher layer.
Further, another method of switching between operating and not operating of the coding section in a higher layer based on a comparison result between the coding result of the lower layer and a predetermined threshold, is proposed (e.g., see Patent Document 2).    Patent Document 1: Japanese Patent Application Laid-Open No. Hei 10-97295    Patent Document 2: Japanese Patent Application Laid-Open No. 2005-80063